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Voice AI That Speaks Carrier-Grade SIP | SignalWire
Enterprise Telephony for Voice AI

Voice AI That Speaks Carrier-Grade SIP

Session Timers, Re-INVITEs, QoS Preconditions. The SIP features every carrier network depends on, built by the team that wrote the SIP stack the industry runs on.

20+
years of production SIP engineering
< 1.2s
typical AI response latency
2,000+
companies in production
2.7B+
minutes processed on the platform
The problem with room-based architectures

Why WebRTC Platforms Break on Real Carriers

Session Timers are mandatory in carrier networks

Periodic SIP re-INVITE or UPDATE messages keep a call session alive. Without them, long-running calls die silently when a network element times out. Room-based media servers have no concept of session timers.

QoS Preconditions guarantee voice quality

IMS networks negotiate quality-of-service parameters before the media session starts. A platform built for WebRTC rooms cannot participate in this negotiation because the protocol has no room equivalent.

Re-INVITE handling is basic telephony

Hold, resume, codec renegotiation, adding or removing media streams, and session refresh all happen through Re-INVITEs. A SIP stack that does not handle Re-INVITE cannot perform basic telephony operations.

SIP dialog state gets lost in translation

When a SIP call becomes a participant in a room, connection-level features disappear. Headers, session parameters, and authentication context get reduced to a room participant object.

Room-based architecture vs. telephony-native

Room-Based Platform

  • SIP features must map to room concepts (most have no equivalent)
  • SIP is a bridge feature; carrier edge cases get secondary attention
  • Requires a separate CPaaS provider for actual phone calls
  • Two platforms, two bills, latency from bridging between them

SignalWire

  • Full SIP dialog state preserved natively
  • Session Timers, QoS Preconditions, Re-INVITEs handled transparently
  • SIP and PSTN are the foundation, not a bolt-on
  • 20+ years of carrier-grade SIP engineering via FreeSWITCH

Build a Voice AI Agent

from signalwire_agents import AgentBase
from signalwire_agents.core.function_result import SwaigFunctionResult

class SupportAgent(AgentBase):
    def __init__(self):
        super().__init__(name="Support Agent", route="/support")
        self.prompt_add_section("Instructions",
            body="You are a customer support agent. "
                 "Greet the caller and resolve their issue.")
        self.add_language("English", "en-US", "rime.spore:mistv2")

    @AgentBase.tool(name="check_order")
    def check_order(self, order_id: str):
        """Check the status of a customer order.

        Args:
            order_id: The order ID to look up
        """
        return SwaigFunctionResult(f"Order {order_id}: shipped, ETA April 2nd")

agent = SupportAgent()
agent.run()

SIP features voice AI agents need

FeatureWhat it doesWhy voice AI needs it
Session TimersPeriodic session refreshLong-running AI calls die without them
QoS PreconditionsPre-call quality negotiationIMS carriers require it before media flows
Re-INVITEMid-call session modificationHold, resume, codec renegotiation
SIP REFERCall transferAI-to-human handoff
SIP SUBSCRIBE/NOTIFYPresence and eventsContact center agent availability
SIP INFOMid-call signalingDTMF, IVR navigation, authentication
SRTP/TLSEncrypted media and signalingHIPAA, PCI, regulatory compliance
Codec negotiationSDP offer/answerInteroperability with any endpoint

Deploy voice AI on carrier infrastructure

1

Define your agent

Write a YAML document or Python class. The same definition works whether the call arrives from an IMS carrier, a SIP trunk, or a PSTN line.

2

Connect your SIP infrastructure

Point your SIP trunks at SignalWire. Session Timers, QoS Preconditions, and Re-INVITEs are handled transparently by the FreeSWITCH-based stack.

3

Test with real carrier traffic

Route test calls through your IMS or enterprise telephony environment. Verify SIP compliance end-to-end.

4

Scale to production

Same platform, same SIP stack, same behavior at carrier scale. No surprises when traffic increases.

SignalWire was built by the team that created FreeSWITCH, the open-source telephony engine that powers major carriers, enterprise contact centers, and unified communications systems worldwide. Production SIP engineering is the foundation, not an add-on.

FAQ

Does SignalWire support IMS carrier deployments?

Yes. The FreeSWITCH-based SIP stack handles Session Timers (RFC 4028), QoS Preconditions (RFC 3312), and the full range of carrier-grade SIP features that IMS networks require.

Can I integrate with existing PBX infrastructure?

Yes. Native SIP trunking, registration, digest authentication, and protocol-level compatibility with session border controllers and enterprise telephony environments.

What about encrypted media and signaling?

SRTP for encrypted media and TLS for encrypted signaling are built into the platform. Required for HIPAA, PCI, and regulatory compliance.

How does transfer work for AI-to-human handoff?

Native SIP REFER for call transfer, with full conversation context (AI summary, session variables, conversation history) carried through to the receiving agent. No webhook gymnastics.

Trusted by 2,000+ companies

Built by the team that wrote the SIP stack.

Deploy voice AI on real carrier infrastructure with full SIP compatibility from day one.